Adaptive dynamic range optimization sound processor

ABSTRACT

In one embodiment, an apparatus for processing sound includes a means ( 401 ) for analysing a sound signal into a number of frequency bands and a means ( 403 ) for applying variable gain to each frequency band independently. Gain in applied under control of a number of gain comparator means ( 409 ) each of which generates a number ot statistical distribution estimates in respect of each signal and compares those estimates to predetermined hearing presponse parameters stored in memory ( 411 ). The numerous gain compensated frequency bands are then combined ( 415 ) in order to generate a single sound signal. The apparatus may be implemented in dedicated hardware embodiment or by software running on a microprocessor.

RELATED APPLICATIONS

This application is a continuation-in-part of application Ser. No.09/478,022 filed Jan. 5, 2000 now U.S. Pat. No. 6,731,767.

FIELD OF THE INVENTION

The present invention relates to the field of devices and methods forprocessing sound and in particular to a processor for improving thespeech perception and comfort of a hearing impaired user. However, whilethe invention is suited for use with hearing impaired people it willalso find application in other communication areas.

BACKGROUND OF THE INVENTION

In general the effects of hearing impairment are characterised by theundesirable conditioning of a sound signal, for example spoken words,along a listener's hearing chain so as to result in attenuation andoften distortion of the signal.

Relatively simple linear gain hearing aids, for example fixed gain aids,have been successful in amplifying sounds to make them audible andrecognisable. One problem with fixed gain aids however is that they areusually not suitable for use over a wide range of sound frequencies andLevels. For example, when using a fixed gain aid the listener oftenfinds that some sounds are inaudible, that is below hearing threshold,while others are at, or above, the loudness discomfort level, (LDL).Such a problem is especially prevalent when the listener is a personwith a narrow dynamic range between the threshold and LDL levels.

Multi-band compression schemes attempt to overcome the problems ofnarrow dynamic range by adapting the gain of the aid in response tochanges in the input sound level within a number of frequency bands,that is, they make use of a non-linear compression scheme. However,non-linear compression schemes introduce distortions into the outputsignals which reduce speech intelligibility. Hearing aids incorporatingmulti-band compression schemes are also difficult to fit and may requirea lengthy investigation of the subject's hearing response.

One type of multi-channel hearing aid is the subject of U.S. Pat. No.5,687,241 to Ludvigsen. In that document there is described amulti-channel hearing aid which splits an input signal into a number ofparallel, filtered channels. The filtered, input signals are eachmonitored by a percentile estimator and on the basis of control signalsgenerated by the percentile estimators the gain of each of the filteredsignals is adjusted. The filtered, gain adjusted signals are thenrecombined, amplified and converted to an acoustic signal.

A problem with the aid of U.S. Pat. No. 5,687,241 is that the percentileestimators must be capable of accommodating large swings in theamplitude of the signal being monitored. Consequently in a digitalimplementation considerable processing power is required in order toundertake the percentile estimation calculations.

A further problem that arises during the operation of multi-channelhearing aids is that fast transient signals having magnitudes exceedingthe maximum comfort level may arise. Typically such transients occur inonly a small number of channels at a particular time however in order toprevent discomfort to the user of the aid the general prior art approachhas been to reduce the total power output of the aid. While such anapproach prevents discomfort it causes undesirable distortion of thesignal in channels unaffected by fast transient signals.

Single channel automatic gain control (AGC) hearing aids operate toreduce the gain at all frequencies in the event that the level of asound should reach a predetermined point. While such hearing aidsprevent the sound from reaching the subject's LDL they also attenuatesome frequency components of the speech signal to such an extent thatthe intelligibility of the speech is reduced.

In summary, prior art hearing aids have associated with their use avariety of problems. Such problems range from inappropriate compressionof signal, which causes undue signal distortion, to onerous processingrequirements which make the aids expensive and difficult to implement.

In light of the prior art it is an object of the present invention toprovide an apparatus which, in the presence of an ambient sound signal,generates a transformed sound signal which conforms to predeterminedamplitude requirements within a range of audible frequencies.

It is a further object of the invention to provide a means whereby fasttransient signals may be suppressed, in order to prevent discomfort tothe user of a multi-channel hearing aid, without introducing signaldistortion into channels unaffected by said transient signals.

SUMMARY OF THE INVENTION

In accordance with one apspect of the present invention, there isprovided an apparatus for processing an ambient sound signal including:

input means for receiving the ambient sound signals;

means for performing a Fourier transform on the input signal andproviding an input spectrum having discrete frequency components eachincluding a coefficient defining the magnitude of the component;

means for multiplying the magnitude coefficients by a predetermined gainvalue and providing magnitude adjusted frequency components;

means for comparing the amplitude of the magnitude adjusted frequencycomponents with predetermined values;

means for attenuating the magnitude of those adjusted frequencycomponents whose magnitude is greater than the predetermined values; and

output means for an output spectrum signal including the frequencycomponents and respective adjusted and attenuated magnitudes.

In accordance with another aspect of the present invention, there isprovided a method for processing an ambient sound signal including thesteps of:

performing a Fourier transform on the ambient sound signal andgenerating an input spectrum having discrete frequency components eachincluding a co-efficient defining the magnitude of the component;

multiplying the magnitude coefficients by a predetermined gain value andproviding magnitude adjusted frequency components;

comparing the amplitude of the magnitude adjusted frequency componentswith predetermined values;

attenuating the magnitude of those adjusted frequency components whosemagnitude is greater than the predetermined values; and

providing an output spectrum signal including the frequency componentsand respective adjusted and attenuated magnitudes.

In accordance with another aspect of the present invention, there isprovided a method for processing an ambient sound signal including thesteps of:

a) performing a frequency analysis on the ambient sound signal togenerate a plurality of analysis signals corresponding to the ambientsound signal;

b) multiplying each of said plurality of analysis signals by acorresponding one of a plurality of gain values to produce a pluralityof magnitude adjusted analysis signals;

c) determining distribution values characteristic of the amplitudedistribution of each of the plurality of magnitude adjusted analysissignals over a period of time;

d) setting said gain values on the basis of comparisons between saiddistribution values and any one or more of a plurality of hearingresponse parameters;

e) processing said plurality of magnitude adjusted analysis signals toform an output signal; and

g) feeding said output signal to a monaural or binaural system havingany one or more selected from the group comprising: a headphone, ahearing aid, a cochlear implant and a mechanical activator driving anossicle in the middle ear of a patient.

In accordance with another aspect of the present invention, there isprovided a method for processing an ambient sound signal including thesteps of:

a) performing a frequency analysis on the ambient sound signal togenerate a plurality of analysis signals corresponding to the ambientsound signal;

b) multiplying each of said plurality of analysis signals by acorresponding one of a plurality of gain values to produce a pluralityof magnitude adjusted analysis signals;

c) determining distribution values characteristic of the amplitudedistribution of each of the plurality of magnitude adjusted analysissignals over a period of time;

d) setting said gain values on the basis of comparisons between saiddistribution values and a plurality of hearing response parameters

e) processing said plurality of magnitude adjusted analysis signals toform an electric output signal; and

f) feeding said output signal to a cochlear implant system.

In accordance with another aspect of the present invention, there isprovided a method for processing an ambient sound signal including thesteps of:

a) performing a frequency analysis on the ambient sound signal togenerate a plurality of analysis signals corresponding to the ambientsound signal;

b) multiplying each of said plurality of analysis signals by acorresponding one of a plurality of gain values to produce a pluralityof magnitude adjusted analysis signals;

c) determining distribution values characteristic of the amplitudedistribution of each of the plurality of magnitude adjusted analysissignals over a period of time;

d) setting said gain values on the basis of comparisons between saiddistribution values and a plurality of hearing response parameters

e) combining said plurality of magnitude adjusted analysis signals toform a combined signal;

f) processing said combined signal to generate an output signal; and

g) feeding said output signal to a mechanical activator driving anossicle in the middle ear of a patient.

In accordance with another aspect of the present invention, there isprovided a method for processing an ambient sound signal for a binauralsystem, including the steps of:

a) performing a frequency analysis on the ambient sound signal togenerate a plurality of analysis signals corresponding to the ambientsound signal;

b) multiplying each of said plurality of analysis signals by acorresponding one of a plurality of gain values to produce a pluralityof magnitude adjusted analysis signals;

c) determining distribution values characteristic of the amplitudedistribution of each of the plurality of magnitude adjusted analysissignals over a period of time;

d) setting said gain values on the basis of comparisons between saiddistribution values and a plurality of hearing response parameters

e) combining said plurality of magnitude adjusted analysis signals toform a combined signal;

f) processing said combined signal to generate a sound output signal;and

g) feeding said sound output signal to a hearing aid, or a headphone, orother electro-acoustic output transformer.

In accordance with another aspect of the present invention, there isprovided an apparatus for processing an ambient sound signal including:

a) a frequency analysis means arranged to generate a plurality ofanalysis signals corresponding to said ambient signal;

b) a magnitude adjustment means coupled to the frequency analysis meansand arranged to adjust the magnitude of each of said analysis signals toproduce a plurality of magnitude adjusted analysis signals;

c) a distribution estimation means responsive to said plurality ofmagnitude adjusted analysis signals and arranged to generatedistribution values characteristic of the amplitude distribution of eachof the said plurality of magnitude adjusted analysis signals over aperiod of time; and

d) a comparison means coupled to the distribution estimation means andarranged to perform comparisons of said distribution values withpredetermined hearing response parameters, said comparison meanscontrolling said magnitude adjustment means on the basis of saidcomparisons;

wherein the magnitude adjustment means, the distribution estimationmeans and the comparison means are implemented by a programmedmicroprocessor coupled to memory storage means, said memory meansstoring the hearing response parameters and include at least one themaximum comfortable levels, optimum audibility levels and thresholdlevels for each of the plurality of frequency components.

In accordance with another aspect of the present invention, there isprovided a computer readable medium, having a program recorded thereon,where the program is configured to cause a computer to execute a methodfor processing an ambient sound signal, said method including the stepsof:

a) performing a frequency analysis on the ambient sound signal togenerate a plurality of analysis signals corresponding to the ambientsound signal;

b) multiplying each of said plurality of analysis signals by acorresponding one of a plurality of gain values to produce a pluralityof magnitude adjusted analysis signals;

c) determining distribution values characteristic of the amplitudedistribution of each of the plurality of magnitude adjusted analysissignals over a period of time;

d) setting said gain values on the basis of comparisons between saiddistribution values and any one or more of a plurality of hearingresponse parameters; and

e) processing said plurality of magnitude adjusted analysis signals toform an output signal.

DESCRIPTION OF THE DRAWINGS

FIG. 1 is an amplitude vs frequency graph including hypotheticalthreshold and loudness discomfort level lines for an unaided, severelyhearing impaired, listener. The shaded regions indicate a hypotheticaldistribution of amplitudes for a speech signal in low background noise.

FIG. 2 is a graph similar to FIG. 1 wherein the speech signal has beenamplified by a linear gain hearing aid.

FIG. 3 is a graph similar to FIG. 1 and FIG. 2 wherein the speech signalhas been processed according to the present invention.

FIG. 4 schematically depicts a hearing aid constructed in accordancewith the present invention.

FIG. 4A schematically depicts a dedicated hardware implementation of ahearing aid constructed in accordance with the present invention.

FIG. 4B schematically depicts, with greater detail, a portion of theapparatus depicted in FIG. 4A.

FIG. 5 is a block diagram of the method of operation of the hearing aidof FIG. 4.

FIG. 6 is a detailed block diagram of the procedure followed at box 515of FIG. 5.

FIG. 7 is a detailed block diagram of the procedure followed at box 517of FIG. 5.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

Referring now to FIG. 1 there is shown a graph with horizontal axisbeing Frequency in Hz and vertical axis being amplitude in dB of soundpressure level (SPL). Plotted on the graph is a speech signal region 1which represents the amplitude and frequency distribution of the speechof a single speaker in a quiet room. Region 1 is demarcated into 5sub-regions bounded by lines 2,4,8,9,10 and 12. The sub-region betweenlines 2 and 4 represents the 90-100th percentile distribution of thesingle speaker speech signal across the 250 Hz-6000 Hz frequency range.Similarly lines 4 and 8 bound the 70th-90th percentile, lines 8 and 9the 30th-70th percentile, lines 9 and 10 the 10th-30th percentile andlines 10 and 12 the 0^(th) to 10^(th) percentile. For furtherinformation regarding such graphs, reference can be made to a paper by HK Dunn and S D White, entitled “Statistical Measurements onConversational Speech”, Journal of the Acoustical Society-of America,11: 278-288, 1940. The paper includes measured amplitude distributionsfor male and female speakers in quiet.

Also plotted on the graph of FIG. 1 is the hearing response 3 of aseverely hearing impaired listener. Hearing response 3 is bounded at itslower border by threshold level 5 and at its upper border by LDL 7.Speech signal frequency components which fall between these two levelswill be perceived by the hearing impaired person while those that fallbelow will not. In the case of FIG. 1 it will be realised that thefrequency components of the speech signal in the range 1000-6000 Hz areall below the threshold of the listener. For example, at 4000 Hz thelistener's threshold level is about 95 dB and the maximum speech levelis about 60 dB. Also plotted on the graph of FIG. 1 is the optimumaudible level 6.

Referring now to FIG. 2 there is depicted once again the frequencydistribution 1 of the speech of a single speaker in a quiet room andalso the hearing response of the severely hearing impaired person ofFIG. 1, this time amplitude distribution is now about 100 dB.Accordingly the top 10% of the speech signal at 4000 Hz is now audible.The remaining 90% of the amplitude distribution of speech at 4000 Hzfalls below the threshold level and is not heard at all. The upper partof the speech amplitude distribution at 1000 Hz lies above the LDL of110 dB SPL and will produce an uncomfortably loud sound unless limitedby an AGC which would reduce the audibility at 4000 Hz and otherfrequencies, or by peak clipping (a form of instantaneous non linearcompression) which would introduce distortion across a wide range offrequencies.

Referring now to FIG. 3 there is depicted a third graph, this timeillustrating the hearing response of the severely hearing impairedsubject of FIG. 1 when wearing an adaptive dynamic range optimisation(ADRO) hearing aid according to the present invention. It will be notedthat the amplitude and frequency distribution 1 of the speech signal nowfalls almost entirely within the boundaries of the acceptable levelswhich may be presented to the hearing aid user, so that all thefrequency components of the speech signal are perceived by the listener.Consequently there is a marked increase in audibility, a markedreduction in distortion, of the signal perceived by the subject, and acorresponding increase in intelligibility of the words comprising thesignal.

At the same time, no frequency component of the output signal exceedsthe listener's LDL. The construction and operational processes of ahearing aid according to the invention will now be explained.

With reference to FIG. 4 there is depicted a digital hardwareimplementation of the hearing aid. Sound waves are transduced bymicrophone 11 and, the electrical signal so produced, conditioned byanalog conditioning module 13. Conditioning module 13 includes standardcircuits for pre-amplifying and low pass filtering the signal prior toits processing by analog to digital converter 15. Analog to digitalconverter 15 produces a 16 bit digital signal which is conveyed tomicroprocessor 17. Microprocessor 17 operates according to a programstored in EPROM 19. The microprocessor performs a fast Fourier transformand generates an input spectrum which is processed, as will bedescribed, to generate an output spectrum comprising a plurality offrequency components.

The output spectrum is then subjected to an inverse fast Fouriertransform in order to produce a digital output signal. The digitaloutput signal is passed to a suitable digital to analog converter 21which generates an analog signal. The analog signal is passed throughsmoothing filter 23 and to power amplifier 25. The amplified signal thendrives earpiece 27.

FIG. 4 a depicts a dedicated hardware implementation of the inventionfor purposes of explanation. While FIG. 4 a illustrates the invention asif individual parts of processor 17 were embodied in dedicated hardware,the invention is most readily implemented by the arrangement of FIG. 4.

Referring to FIG. 4 a, the signals from the ADC 15 are subjected tomulti-channel frequency analysis and classified into n (for example n=8)frequency bands in analysis section 401. The frequency analysis depictedis carried out by means of n band-pass switched-capacitor filters.

The resulting n frequency analysis signals from frequency analysissection 401 are then conveyed to magnitude adjustment section 403. Themagnitude of each of the n signals is adjusted by one of n gain controlelements 405-407 under the control of a gain computation section 409comprising n gain computation elements 410,412,414. Each of the n gaincomputation elements monitors a corresponding one of the n gain adjustedsignals, processes its signal in a manner that will shortly be explainedwith reference to FIG. 4 b, and controls the amount of gain applied bygain control elements 405-407 of magnitude adjustment section 403. Map41 comprises a memory storing a set of previously determined hearingresponse parameters. The predetermined parameters are the thresholdlevel (TL), the maximum comfortable level (MCL), maximum power outputlevel (MPO), optimal audible level (OPT), and maximum gain level(MAXGAIN) for the intended user of the device at each of the centrefrequencies of the n channels. The maximum gain level is just below thelevel at which feedback occurs for the channel in question duringoperation of the aid. The maximum gain level is determined duringfitting of the aid.

The magnitude adjusted analysis signals are passed to maximum poweroutput limiting section 413 comprised of n maximum power output limiterswhich compare each of the n signals with the corresponding predeterminedmaximum power output level stored in map 411. The MPO limiters ensurethat the signal cannot exceed the predetermined MPO value for eachchannel. The output limiters are designed to act within 0.1 millisecondsin order to suppress fast transients. It will be noted that the MPOlimiters act independently so that a signal in a particular channel onlyaffected if a fast transient has occurred in that channel. The n signalsthen pass to reconstruction stage 415 which recombines the n magnitudeadjusted signals, typically by summing the waveforms from each channel.

The signal processing operation of the first gain computation element ofmagnitude adjustment stage 409 will now be explained in greater detailwith reference to FIG. 4 b which illustrates the internal configurationof first gain computation element 410. The other gain computationelements are similarly arranged. It will be seen that the signalemanating from the adjustment element 405 is monitored by threepercentile estimators 431-433. The percentile estimators each generate aceiling value signal which indicates the level that the signal beingmonitored falls beneath for a particular percentage of the monitoringperiod. In the present embodiment the percentile estimators 431-433 areset to produce estimates of the ceiling values reached by the monitoredsignal 98% of the time, 70% of the time and 30% of the timerespectively. As can be seen from FIGS. 1-3, when the monitored signalis derived from speech, the value that the signal falls beneath 98% ofthe time is much greater than the value that it falls beneath 30% of thetime. The design of percentile estimator hardware is explained in U.S.Pat. No. 4,204,260 incorporated herein by reference.

The percentile level estimate signals generated by percentile estimators431-433 are passed to comparators 435-437. Comparator 435 compares the98th percentile estimate with the maximum comfortable level in respectof channel 1 which is stored in map 411. Similarly comparators 436 and437 compare the 70th and 30th percentile estimates with thepredetermined optimum audibility and threshold levels stored in map 411.The outcomes of the comparisons are conveyed to gain adjust unit 439.The gain adjust unit 439 is typically implemented as a programmablelogic array that would control the gain of the amplifier 405 accordingto the following logic. In the event that the 98th percentile estimateexceeds the maximum comfortable level then the gain should reduceslowly. Otherwise, if the 70^(th) percentile estimate is below theoptimum audibility level, the gain should increase slowly until the gainis equal to the corresponding MAXGAIN level stored in Map 411 or the98th percentile estimate reaches the maximum comfortable level.Otherwise, if the 30th percentile estimate is above the threshold level,then the gain should fall slowly.

The rates of rise and fall of the gain control unit 405 are typically 3to 10 dB per second. The level of gain to be applied is transmitted fromgain adjust section 439 to gain adjustment element 405 and the magnitudeof the signal is adjusted accordingly.

While the above description explains a dedicated hardware implementationof the invention, as previously explained it will be most convenient toimplement the invention by means of an appropriately programmed digitalsignal processor integrated circuit as illustrated in FIG. 4.

The software for programming the digital signal processor EPROM 19 ofFIG. 4 may be stored in a computer readable medium, including thestorage devices described below, for example, before being loaded fromthe computer readable medium to the DSP chip.

In some instances, the software may be encoded on a CD-ROM or floppydisk. Alternatively the software may be read from a network via a modemdevice. Still further, the software can be loaded into the DSP chip fromother computer readable medium including magnetic tape, a ROM orintegrated circuit, a magneto-optical disk, a radio or infra-redtransmission channel, a computer readable card such as a PCMCIA card,and the Internet and Intranets including e-mail transmissions andinformation recorded on Websites and the like. The foregoing is merelyexemplary of relevant computer readable mediums. Other computer readablemedia may be practiced.

Referring now to FIG. 5 a block diagram of the procedural steps followedby the program stored in EPROM 19 of FIG. 4 is presented.

At box 503 microprocessor 17 performs a fast Fourier transform upon thedigital signal output of ADC 15. The fast Fourier transform produces aninput spectrum consisting of N magnitude and N phase components whichare stored in volatile memory, at box 505. Typically, N would take thevalue of 32, 64 or 128.

At box 507 each of the N magnitude components is multiplied by acorresponding one of N gain values. The results of the multiplicationsundertaken at box 507 are stored in volatile memory at box 509.

At box 515 each one of the N 30th, 70th and 98th percentiles of thedistributions over time of the magnitudes of the frequency components iscompared to the corresponding one of the N magnitude estimates of theoutput spectrum. The estimates are adjusted on the basis of thecomparisons as will be explained with reference to FIG. 6.

At box 517 the percentile estimates are compared with valuescharacteristic of a particular hearing response and on the basis of thecomparisons the gain values at each frequency, are adjusted as will belater described in reference to FIG. 7.

At box 511 the magnitude of each of the N frequency components iscompared to a predetermined maximum power output level (MPO) for thatparticular frequency component. If the magnitude of a frequencycomponent is found to be greater than the MPO at the given frequencythen it is set equal to the MPO level. This operation is designed toprevent fast transient signals of a certain frequency from rising abovethe LDL at that frequency without affecting signals at otherfrequencies. The MPO values are set during fitting of the ADRO hearingaid to suit the individual listener's hearing response. The MPO valuesused do not have to be the same as the LDL values, which are alsopredetermined during fitting, although they will usually be similar invalue.

At box 513 an inverse fast Fourier transform is performed on the Nmagnitude and N phase components in order to reconstitute a digital timedomain signal for subsequent processing by digital to analog converter21.

Referring now to FIG. 6 there is depicted in detail a flowchart of theprocedural steps required to implement box 515 of FIG. 5.

Before explaining the procedural steps in the flowchart the followingvariables, which appear in FIG. 6 and FIG. 7 will be defined.

TL

: a one dimensional array for holding N threshold level values.

MCL

: a one dimensional array for holding N maximum comfortable levels. TheMCL would usually be set just below the LDL at each frequency.

Opt

: a one dimensional array for holding N optimal audible levels. Opt [n]would typically be set halfway between TL [n] and MCL [n] at eachfrequency.

X30

: a one dimensional array for representing the estimate of the 30thpercentiles of the amplitude distributions of each of the N differentfrequency components.

X70

: a one dimensional array for holding the estimates of the 70thpercentiles of the amplitude distributions of each of the N frequencycomponents.

98

: a one dimensional array for holding the estimates of the 98thpercentiles of the amplitude distributions of each of the N frequencycomponents.

Gain

: a one dimensional array for holding the N gain values, one for each ofthe N frequency components.

GainUp: a variable for holding the magnitude of the step by which avalue stored in Gain

is to be increased.

GainDown: a variable for holding the magnitude of the step by which avalue stored in Gain

is to be decreased.

n: a counter variable for indexing a particular one of the N frequencycomponents.

EstUp30: a variable for holding the magnitude of the step by which a X30

value is to be increased.

EstDown30: a variable for holding the magnitude of the step by which aX30

value is to be decreased.

EstUp70, EstDown70, EstUp98, EstDown98: corresponding variables for the70th and 98th percentile estimates.

OutSpec

: a one dimensional array holding the magnitudes of the N frequencycomponents of the output spectrum of box 509 of FIG. 5.

Returning now to FIG. 6 at box 603 counter n is set to 1. At box 611,the magnitude of the output spectrum at frequency n is compared with theestimate of the 30th percentile of the amplitude distribution atfrequency n. If the magnitude is greater than or equal to the estimated30th percentile, the estimate is increased by an amount EstUp30 at box613, otherwise the estimate is decreased by an amount EstDown30 at box615. The ratio of the step sizes EstUp to EstDown is equal to i/(100−i)where i is the required percentile. Thus for the 98th percentile (i.e.i=98), the EstUp step is 49 times the EstDown step.

For the 70th percentile (when i=70) the EstUp and EstDown steps in theratio of 7:3. For the 30th percentile (when i=30) the EstUp and Est Downsteps are in the ratio of 3:7. After repeated iterations through theprocess of FIG. 5, the estimates of the percentiles will stabilise atappropriate values.

For example, at the 98th percentile, large upward steps which occur 2%of the time will be balance by downward steps that are 49 times smallerbut occur 49 times more frequently. By varying the TotalSstepSsize(which is equal to EstUp+EstDown) the maximum adaptation rate of theestimates can be controlled. Boxes 617 to 627 are used to estimate the70th and 98th percentiles of the amplitude distribution in a manneranalogous to the 30th percentile. The frequency counter is incrementedat box 629. Box 631 transfers control back to the main process when thepercentile estimates for each frequency have been updated.

Next referring to FIG. 7, the steps in adjusting the gain for eachfrequency are described. As in FIG. 6, variable n is used to stepthrough the frequencies one at a time. If the 98th percentile estimateis greater than the maximum comfortable level at the given frequency,then the gain at the given frequency is reduced by one GainDown step atbox 707. At box 709 the 70th percentile estimate of the currentfrequency components is compared with the optimum audibility level ofthe frequency components. If the 70th percentile is below the Opt valuefor the current frequency component, then the Gain at the givenfrequency is increased by one GainUp step at box 713. Alternatively ifthe tests at both box 705 and 709 are negative then control diverts todecision box 711. If the 30th percentile estimate exceeds the optimumaudibility level value then control flows to box 707 where the gain atthe frequency is reduced by one GainDown step. Box 715 tests the gainvalue for the current frequency component to see if it is so high thatfeedback is likely to occur. If the result of the test at box 715 ispositive then the gain value for the current frequency is set to thehighest value that does not cause feedback to occur at box 717. Controlthen flows to box 719 at which the frequency counter n is incrementedand then to box 721 at which point the entire process is repeated withrespect to the next frequency component.

It will be noted that the above procedure adjusts the gain valuesindependently of the actual values of the input spectrum. As a resultmicrophone 11 could be replaced with another microphone of arbitraryfrequency response, an induction loop, an RF microphone or a directconnection to a telephone, or other electronic device, without requiringreadjustment of the aid's operating parameters relating to thelistener's hearing such as the LDL and threshold levels.

Furthermore, it will be understood that the invention acts to reduce thedynamic range of the components of the output spectrum relative to thoseof the input spectrum. For example, input acoustic signals may typicallyvary over a range of 100 dB in which case 19 bit arithmetic would berequired to digitally implement percentile estimators responsive to theinput spectrum. In contrast, the output signal for severely-hearingimpaired users will only vary over about 50 dB so that advantageouslyonly 10 bit arithmetic is required to implement percentile estimators inthe arrangement of the present invention.

One parameter that must generally be adjusted in the event that themicrophone is exchanged is the set of values stored in the MaxGain

array. The reason for this is that the gain depends on the differencebetween the input and output signals and not simply on the outputsignal. Feedback problems are unlikely to occur where the inputmicrophone is acoustically well isolated from the output as is the casefor an induction loop or telephone coil.

While the above system has been described with respect to a limitednumber of embodiments it will be realised that variations are possible.For example the output spectrum, digital output signal or correspondinganalog output, generated during the various stages of processing couldbe passed directly to a cochlear implant processor, or digital soundprocessor, in which case the present invention would operate as thefront-end of a further signal processor. An example of the generaloperation of a cochlear implant system is described in U.S. Pat. No.4,532,930, the contents of which are incorporated herein by reference.

For hearing aids, headphones, and middle ear transducers, the signalfrom each of the FFT channels is processed by the ADRO Rules, includingmultiplication by the gain and maximum power output limiting. Theprocessed channels are then recombined to produce a single signalchannel using the inverse FFT.

In an alternative arrangement, a plurality of output signals can beapplied to the electrodes of a cochlear implant.

In one arrangement, the FFT channels are combined to produce, forexample, 22 channels before the ADRO processing is applied to each oneof the combined channels. The ADRO processing refers to the gainmultiplication and the maximum power output limiting.

Alternatively, the ADRO processing is applied to each one of the FFTchannels, followed by a combination or selection of channels to producethe individual channels for the cochlear implant stimulation.

The final steps for the cochlear implant do not include the inverse FFTto produce a single output channel. Instead, the plurality of channeloutputs are selectively coded as electrical signals to be applied to aplurality of electrodes, producing direct electrical excitation ofauditory nerve fibres and the desired loudness perceptions in thedefined output dynamic range at each stimulated electrode.

In the other cases (headphones, hearing aid, electromechanicaltransducer), the analog electrical output signal is applied directly tothe input contacts of an electro-acoustic or electro-mechanicaltransducer which converts the electrical signal to an acoustic ormechanical vibration which is then transmitted to the inner ear by theusual means where it is processed by the usual hearing mechanisms.

The method according to this disclosure can be used to optimisecombinations of acoustic and electric output signals either in abinaural bimodal device with acoustic signal in one ear and electricsignal in the other or monaural hybrid device with electric and acousticsignals in the same ear at the same time.

An important feature inherent to the method according to this disclosureis that the phase of each of the FFT components is maintained during thelimiting step. This minimises the distortion of the waveform afterlimiting of one or more components.

Both bimodal and hybrid stimulation perform better when the signals arepresented with a common amplitude envelope (promoting fusion of theinformation from the two signals into a single perceptual stream), andat comparable loudness. Experimental and theoretical considerations ofthis point are covered by Blamey et al, Ear & Hearing 21, 6-17, 2000“Monaural and binaural loudness measures in cochlear implant users withcontralateral residual hearing.” Subsequent clinical trials with thisprocessing in cochlear implants (James et al, Ear & Hearing 23, 498-588,2002) and hearing aids have shown improved speech perception, comfortand sound quality compared with alternative amplification schemes. Thebenefits come from individual optimization of output levels andinformation content in the plurality of frequency channels used in theprocessing. Provided that the matching and control of loudness isconsistent across the output transducers, and across ears, as well asacross frequency channels, these benefits will be maintained forcombinations of electrical, mechanical, and acoustic output signals.

Some configurations of hearing loss make various combinationsadvantageous to individual listeners. For example, binaural fitting ofheadphones is most advantageous to a listener with normal hearing. For aperson who has a severe hearing loss in both ears but does not wantbinaural surgery, or a person who has a moderate hearing loss in one earand a total hearing loss in the other, a bimodal fitting of a hearingaid and a cochlear implant may be most advantageous. Hybrid fitting of acochlear implant and hearing aid in one ear is appropriate for a personwith good low frequency residual hearing and very poor high frequencyhearing. Use of middle ear electromechanical transducers may beadvantageous for some listeners with ossicular or tympanic membranedamage, or who are using a totally implanted hearing aid.

The invention might also be applied to ear muffs or hearing protectorsin order to help people with normal hearing communicate in the presenceof loud background noises such as hums, whistles and some types ofstatic. Such noises are said to be stationary and have a narrow dynamicrange so that their Low, Mid and High percentiles are close together.The Low percentile is constrained to lie below the threshold accordingto the above described embodiment of the invention.

Accordingly, the background noise is reduced to a low level. If thebackground noise is also characterised by having a narrow frequencyrange, such as a whistle, then a device according to the invention canbe set to remove the noise from the output signal, while keeping most ofthe other (dynamic) spectral details unaffected.

Telephone and radio communication systems also have requirements foraudibility and maximum power output levels that can be expressedsimilarly to the needs of hearing aid users. The present invention canbe used to ensure that these requirements are met by an appropriatechoice of the processing parameters. In these cases the threshold andLDL parameters would be determined for normal listeners and would notneed to be adjusted on an individual basis. The invention can also beused to optimize signals prior to further analysis by, for example, anautomatic speech recognition system.

Accordingly the following claims are to be constructed broadly and arenot intended to be limited to the previously described preferredembodiment.

1. An apparatus for processing an ambient sound signal including: inputmeans for receiving the ambient sound signals; means for performing aFourier transform on the input signal and providing an input spectrumhaving discrete frequency components each including a coefficientdefining the magnitude of the component; means for multiplying themagnitude coefficients by a predetermined gain value and providingmagnitude adjusted frequency components; means for comparing theamplitude of the magnitude adjusted frequency components withpredetermined values; means for attenuating the magnitude of thoseadjusted frequency components whose magnitude is greater than thepredetermined values; and output means for an output spectrum signalincluding the frequency components and respective adjusted andattenuated magnitudes.
 2. The apparatus according to claim 1, whereinthe means for performing a Fourier transform, the means for multiplyingthe magnitude coefficients, the means for comparing the amplitude, andthe means for attenuating the amplitude are implemented by a programmedmicroprocessor coupled to memory storage means.
 3. The apparatusaccording to claim 2, wherein the predetermined values are based uponhearing response parameters.
 4. The apparatus according to claim 3,wherein the hearing response parameters comprise any one or more ofloudness discomfort levels, maximum comfortable levels, comfortablelevels, optimum audibility levels, and threshold levels for each of theplurality of frequency components.
 5. The apparatus according to claim1, further including means to perform an inverse Fourier transform onthe output spectrum signal.
 6. The apparatus according to claim 5,further including a digital to analogue converter to convert the outputof the inverse Fourier transform to an analogue signal.
 7. The apparatusaccording to claim 1, incorporated as the front-end of a further signalprocessor.
 8. The apparatus according to claim 1, wherein themicroprocessor is programmed to calculate and store in memory,distribution values indicative of the distribution of the magnitude ofeach of said plurality of adjusted frequency components over a period oftime.
 9. The apparatus according to claim 8, wherein the microprocessoris programmed to determine and store in memory, one or more distributionvalues which are approximately the 30th, 70th, 90th and 98th percentilesof the magnitude of each of said plurality of adjusted frequencycomponents over a period of time.
 10. The apparatus according to claim1, wherein the means for attenuating includes a plurality of limitingmeans responsive to said magnitude adjusted analysis signals andarranged to limit the power of each of said signals to below acorresponding plurality of predetermined levels.
 11. An apparatus forprocessing an ambient sound signal including: a) a frequency analysismeans arranged to generate a plurality of analysis signals correspondingto said ambient signal; b) a magnitude adjustment means coupled to thefrequency analysis means and arranged to adjust the magnitude of each ofsaid analysis signals to produce a plurality of magnitude adjustedanaalysis signals; c) a distribution estimation means responsive to saidplurality of magnitude adjusted analysis signals and arranged togenerate distribution values characteristic of the amplitudedistribution of each of the said plurality of magnitude adjustedanalysis signals over a period of time; and d) a comparison meanscoupled to the distribution estimation means and arranged to performcomparisons of said distribution values with predetermined hearingresponse parameters, said comparison means controlling said magnitudeadjustment means on the basis of said comparisons; wherein the magnitudeadjustment means, the distribution estimation means and the comparisonmeans are implemented by a programmed microprocessor coupled to memorystorage means, said memory means storing the hearing response parametersand include at least one the maximum comfortable levels, optimumaudibility levels and threshold levels for each of the plurality offrequency components.